Replace BundleFilter with RtpDemuxer in RtpTransport.
BundleFilter is replaced by RtpDemuxer in RtpTransport for payload
type-based demuxing. RtpTransport will support MID-based demuxing later.
Each BaseChannel has its own RTP demuxing criteria and when connecting
to the RtpTransport, BaseChannel will register itself as a demuxer sink.
The inheritance model is changed. New inheritance chain:
DtlsSrtpTransport->SrtpTransport->RtpTranpsort
The JsepTransport2 is renamed to JsepTransport.
NOTE:
When RTCP packets are received, Call::DeliverRtcp will be called for
multiple times (webrtc:9035) which is an existing issue. With this CL,
it will become more of a problem and should be fixed.
Bug: webrtc:8587
Change-Id: Ibd880e7b744bd912336a691309950bc18e42cf62
Reviewed-on: https://webrtc-review.googlesource.com/65786
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22867}diff --git a/pc/rtptransport_unittest.cc b/pc/rtptransport_unittest.cc
index 0425942..97ea2e4 100644
--- a/pc/rtptransport_unittest.cc
+++ b/pc/rtptransport_unittest.cc
@@ -250,49 +250,37 @@
EXPECT_EQ(1, observer.rtp_transport_sent_count());
}
-class SignalCounter : public sigslot::has_slots<> {
- public:
- explicit SignalCounter(RtpTransport* transport) {
- transport->SignalReadyToSend.connect(this, &SignalCounter::OnReadyToSend);
- }
- int count() const { return count_; }
- void OnReadyToSend(bool ready) { ++count_; }
-
- private:
- int count_ = 0;
-};
-
TEST(RtpTransportTest, ChangingReadyToSendStateOnlySignalsWhenChanged) {
RtpTransport transport(kMuxEnabled);
- SignalCounter observer(&transport);
+ TransportObserver observer(&transport);
rtc::FakePacketTransport fake_rtp("fake_rtp");
fake_rtp.SetWritable(true);
// State changes, so we should signal.
transport.SetRtpPacketTransport(&fake_rtp);
- EXPECT_EQ(observer.count(), 1);
+ EXPECT_EQ(observer.ready_to_send_signal_count(), 1);
// State does not change, so we should not signal.
transport.SetRtpPacketTransport(&fake_rtp);
- EXPECT_EQ(observer.count(), 1);
+ EXPECT_EQ(observer.ready_to_send_signal_count(), 1);
// State does not change, so we should not signal.
transport.SetRtcpMuxEnabled(true);
- EXPECT_EQ(observer.count(), 1);
+ EXPECT_EQ(observer.ready_to_send_signal_count(), 1);
// State changes, so we should signal.
transport.SetRtcpMuxEnabled(false);
- EXPECT_EQ(observer.count(), 2);
+ EXPECT_EQ(observer.ready_to_send_signal_count(), 2);
}
// Test that SignalPacketReceived fires with rtcp=true when a RTCP packet is
// received.
TEST(RtpTransportTest, SignalDemuxedRtcp) {
RtpTransport transport(kMuxDisabled);
- SignalPacketReceivedCounter observer(&transport);
rtc::FakePacketTransport fake_rtp("fake_rtp");
fake_rtp.SetDestination(&fake_rtp, true);
transport.SetRtpPacketTransport(&fake_rtp);
+ TransportObserver observer(&transport);
// An rtcp packet.
const char data[] = {0, 73, 0, 0};
@@ -312,10 +300,14 @@
// handled payload type is received.
TEST(RtpTransportTest, SignalHandledRtpPayloadType) {
RtpTransport transport(kMuxDisabled);
- SignalPacketReceivedCounter observer(&transport);
rtc::FakePacketTransport fake_rtp("fake_rtp");
fake_rtp.SetDestination(&fake_rtp, true);
transport.SetRtpPacketTransport(&fake_rtp);
+ TransportObserver observer(&transport);
+ RtpDemuxerCriteria demuxer_criteria;
+ // Add a handled payload type.
+ demuxer_criteria.payload_types = {0x11};
+ transport.RegisterRtpDemuxerSink(demuxer_criteria, &observer);
// An rtp packet.
const rtc::PacketOptions options;
@@ -324,6 +316,31 @@
fake_rtp.SendPacket(rtp_data.data<char>(), kRtpLen, options, flags);
EXPECT_EQ(1, observer.rtp_count());
EXPECT_EQ(0, observer.rtcp_count());
+ // Remove the sink before destroying the transport.
+ transport.UnregisterRtpDemuxerSink(&observer);
+}
+
+// Test that SignalPacketReceived does not fire when a RTP packet with an
+// unhandled payload type is received.
+TEST(RtpTransportTest, DontSignalUnhandledRtpPayloadType) {
+ RtpTransport transport(kMuxDisabled);
+ rtc::FakePacketTransport fake_rtp("fake_rtp");
+ fake_rtp.SetDestination(&fake_rtp, true);
+ transport.SetRtpPacketTransport(&fake_rtp);
+ TransportObserver observer(&transport);
+ RtpDemuxerCriteria demuxer_criteria;
+ // Add an unhandled payload type.
+ demuxer_criteria.payload_types = {0x12};
+ transport.RegisterRtpDemuxerSink(demuxer_criteria, &observer);
+
+ const rtc::PacketOptions options;
+ const int flags = 0;
+ rtc::Buffer rtp_data(kRtpData, kRtpLen);
+ fake_rtp.SendPacket(rtp_data.data<char>(), kRtpLen, options, flags);
+ EXPECT_EQ(0, observer.rtp_count());
+ EXPECT_EQ(0, observer.rtcp_count());
+ // Remove the sink before destroying the transport.
+ transport.UnregisterRtpDemuxerSink(&observer);
}
} // namespace webrtc
